In order to precisely separate sound waves of a target sound source, and to suppress target external sound like noises, in general, it is necessary to apply a directional microphone, and to dispose the plural directional microphones side by side at equal to or wider than a certain pitch. However, in the case of a compact sound collector device like an IC recorder, it is difficult to apply the sound collecting technology of employing a directional microphone and of utilizing the plural microphones with a wide pitch. In addition, a precise sound-source separation by an application of such a sound collecting technologies to recorded sound from plural sound sources and having undergone an artificial down-mix process is also difficult.
Hence, a large number of technologies of analyzing an amplitude difference and a phase difference between signals output by respective microphones after recording of sound wave, and performing a signal processing in accordance with an analysis result, thereby separating and extracting a target sound source have been proposed. In recent years, a statistical analysis, a frequency analysis, a complex analysis, etc., are applied to detect a difference in waveform structure of input signals, and the detection result is utilized for a sound-source separation process.
For example, a signal processing such that a conversion from a time axis to a frequency axis is performed on an input signal, a phase difference for each frequency is calculated, a frequency band of an input sound wave from a target sound source is specified based on the calculated difference, and the sound wave within that frequency band is emphasized is performed (see Patent Document 1).
In addition, in the signal processing, it is determined whether or not an input sound wave is in a target direction based on input signals from two microphones closely disposed to each other, a phase difference between the two input signals is corrected, thereby emphasizing sound present in the target direction (see Patent Document 2). The two input signals are referred to each other, and a filter is sequentially updated based on an obtained signal (see Patent Document 3).